Packet loss anticipation and pre emptive retransmission for low latency media applications

ABSTRACT

In many low latency media applications it is important to transmit media data packets from a media source to one or more media destinations as promptly as possible, while also ensuring that all media data packets that may be lost due to transmission errors are retransmitted and received correctly at the media destination. This invention described a system to do this with a system and methods for anticipating media data packet loss and making preemptive media data packet retransmission requests by dynamically computing a metric and decision logic for retransmission request that includes a need based factor from the media consuming application.

CROSS REFERENCE TO RELATED APPLICATIONS

This application claims priority under 35 U.S.C. §119(e) to U.S.Provisional Patent Application Serial No. 61/512,924, filed Jul. 29,2011, entitled “Techniques for broadcasting media over a local networkto multiple destinations” the entire content of which is incorporatedherein by reference.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention is directed to network communications and todigital media sourcing, transmission and rendering.

2. Summary of the Invention

The present invention is directed to low latency media applicationswhere it is important to transmit media data packets from a media sourceto one or more media destinations as promptly as possible, while alsoensuring that all media data packets that may be lost due totransmission errors are retransmitted and received correctly at themedia destination. This invention includes a system and methods foranticipating media data packet loss and making preemptive media datapacket retransmission requests of the media source.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 illustrates an overview of the devices in a system in accordancewith one embodiment.

FIG. 2 illustrates a schematic of the devices in a system in accordancewith one embodiment.

FIG. 3 illustrates a typical TCP/IP connection from a media source to amedia rendering device.

FIG. 4 illustrates the TCP/IP sliding window protocol.

FIG. 5 illustrates typical TCP/IP packet communication between sourceand destination.

FIG. 6A illustrates the number of packets received at each packet topacket delay.

FIG. 6B illustrates the percent of total packets received for eachpacket to packet delay.

FIG. 7 illustrates packet communication between source and destinationwith a pre emptive retransmission request.

FIG. 8 illustrates the transmission elements used in the LAP methods ofthis invention.

FIG. 9 illustrates the destination components used to make pre emptiveretransmission request.

FIG. 10 illustrates a detailed example of the received packet collectionused at the destination.

FIG. 11 illustrates the LAP algorithm used at the destination.

FIG. 12 shows these calculations computing the metrics M_(x).

FIG. 13 shows a plot of the calculations shown in FIG. 12.

DETAILED DESCRIPTION OF THE INVENTION

Today there are many forms of digital media, many types of digital mediasources, many types of digital media playback (rendering) systems andlots of ways of connecting media sources to media playback systems.

Digital media, hereafter referred to as media, comes in many forms,formats and containers, including Digital Video Disks, media files andmedia streams. The media contents can be audio, video, images or metadata media components and various combinations of each. For example apopular audio format is known as MP3 and a popular video format is H264.MP3 is an audio-specific media format that was designed by the MovingPicture Experts Group (MPEG) as part of its MPEG-1 standard and laterextended in the MPEG-2 standard. H264 is a standard developed by theInternational Organization for Standardization (ISO)/InternationalElectrotechnical Commission (IEC) joint working group, the MovingPicture Experts Group (MPEG). Movies are typically multimedia formatswith a video and multiple audio channels in it. For example a 5.1 moviecontains 1 video channel (media component) and 6 audio channels (audiocomponents). 5.1 is the common name for six channel surround soundmultichannel audio systems.

Digital media sources include media devices such as Digital Video Diskplayers, Blu-ray players, computer and mobile devices, and internetbased “cloud” media services. Blu-ray Disc (BD) is an optical discstorage medium developed by the Blu-ray Disc Association. Internet basedmedia services include services such as Netflix and Spotify™. Netflix isa media service and trademark of Netflix Inc. Spotify is a media serviceand trademark of Spotify Ltd. Digital media playback (media renderingdestinations) systems include computer based devices, laptops andsmartphones, as well as network audio and video devices. A SmartTV is anexample of a digital media rendering device that can play media from aninternet (cloud) based media service such as Netflix™. A SmartTV, whichis also sometimes referred to as “Connected TV” or “Hybrid TV”, is usedto describe the integration of the internet and Web features into moderntelevision sets and set-top boxes, as well as the technologicalconvergence between computers and these television sets/set-top boxes.An internet radio device is another example of a digital media renderingdevice.

The connectivity between these media sources and devices is varied, butis evolving over time towards network based connectivity using IPprotocols. This is because IP connectivity is convenient, ubiquitous andcheap. IP stands for Internet Protocol. An IP networked device is adevice that adheres to the Internet Protocol suite standard. TheInternet Protocol suite is defined by the Internet Engineering TaskForce [IETF] standards body. The Internet is a global system ofinterconnected computer networks that use the standard Internet Protocol(IP) suite.

IP networks come in many forms; the most prevalent being Ethernet basedwired IP networking. Ethernet is a family of computer networkingtechnologies for local area networks (LANs) that is standardized as IEEE(Institute of Electrical and Electronics Engineers) Standard 802.3. Inrecent years with the prevalence of mobile computing devices, Wi-Fi hasbecome the most popular means for connecting network devices wirelessly.Wi-Fi is a trademark of the Wi-Fi Alliance and a brand name for productsusing the IEEE 802.11 family of standards. A Wi-Fi network is a type ofIP network.

The convenience and benefits of IP networking means that all of thesemedia sources and playback systems, if not already network enabled, arebecoming network enabled. Many Blu-ray players now have Ethernet andWi-Fi network connectivity. Today most higher end TVs are smart TVs thathave network capability. Similarly audio play back devices and evenradios are network and internet enabled.

Mobile devices, such as mobile phones, tablets, readers, notebooks etc,are able to receive and store media and have powerful media (audio andvideo) capabilities and are connected to the internet via cell phonedata services or broadband links, such as Wi-Fi that are high bandwidthand can access online media services that have wide and deep content.

The use cases or applications of these various forms of digital media,media services and media sources and playback systems have beenevolving. Initially it was enough to connect a media source to a mediadestination over an IP network. This is widely used today with Internetbased media source services, such as Netflix and a computer as a mediadestination. Users watch Netflix movies streamed over a wired IP network(the internet) to a computer. This is a case of a single point (one IPsource) to single point (one IP destination) connection over a wired IPnetwork. Even though the Netflix media service may send the same mediato multiple households, each of these is a single point to single pointconnection TCP/IP connection. A further evolution of this is to use awireless, Wi-Fi connection, instead of a wired Ethernet connection. Thisis still a single point to single point connection.

The applications targeted in this invention are for a further extensionof the above use cases where the media source connects to multipledestinations rather than a single destination. These are single point(one IP source) to multi point (multiple IP destinations) applications.An example would be where a user is playing a 5.1 movie media file to awireless video playback device and 6 independent wireless audiodestinations making up a full 5.1 surround sound system. In this casethe media is going from one media source to 7 media destinationssimultaneously. In another example, a user is playing music from onemedia source to 6 audio playback systems placed around the home in 6different rooms.

In both of these cases, it is necessary to play (render) the media atall destinations time synchronously. Furthermore, it is necessary tolimit the use of resources at the media source, such as keeping memoryuse to a minimum. In addition, it is necessary with multiple devicesreceiving media to manage network bandwidth efficiently.

In some applications, the video media may be rendered through one path,for example a specialized hardware path, and the audio may be renderedthrough a different network path. When different media components of thesame media are going through different paths, it is necessary to keeppath delays (path latency) to a minimum. This is necessary to keep thedifferent media components time synchronized. In these applications,keeping media network transport latencies to a minimum is important.

Furthermore, when the network is Wi-Fi, network packet losses can behigh and it is necessary to mitigate these in order to deliveruninterrupted playback.

The general structure of these application are that of multiple IPnetworked media source devices choosing, connecting and playing media toone or more IP networked media playback devices over an IP communicationnetwork.

FIG. 1 shows an exemplary system 100 having multiple media sourcedevices 104 and multiple media destination devices 106.

FIG. 2 is a schematic diagram of such a media system 100 with one ormore IP network-enabled media source devices 104 and one or more IPnetwork enabled media destination devices 106 connected via an IPnetwork 120.

Referring to both FIG. 1 and FIG. 2, a media source device 104 can beany variety of computing devices that can originate digital mediaincluding computers (e.g. desktop, notebook 14, tablet 12, handheld),mobile devices (e.g. smart phone 10, electronic book reader, organizerdevices), as well as set-top boxes and game machines 16. The media isany form of digital media, including audio or video, images, data,and/or Meta data.

Media destination devices 106 are devices that can receive digital mediaover an IP network 120 and play this media. This includes IP-enabledaudio and/or video and/or imaging devices that can render audio or videoor images or combinations of these at the same time. Media destinationdevices 106 include computers (e.g. desktop, notebook 15, tablet 13,handheld), mobile devices (e.g. smartphones, tablets, notebooks 15),network enabled TVs 20, network enabled audio devices 18, 22. If themedia is audio, playing the media means rendering the audio such that auser can listen to the audio. If the media is video, playing meansrendering the video such that a user can view the media. If the mediaincludes both audio and video, it means rendering both the audio and thevideo. If the media is images, playing means displaying these images ona screen. In this description, media destination devices 106 may also bereferred to as media renderers or combinations of these terms.

In the media environment 100 of the present invention, each media source104 can send its media to a selected set of media destination devices106 for playback.

The network 120 and all networks used and described in this invention toconnect all devices, including the media sources 104 with the mediadestinations 106 may be any network that supports an IP protocol. Thisincludes any wired IP connectivity mechanism including Ethernet if wiredand if wireless it includes any wireless IP connectivity mechanismincluding Wi-Fi. If this 120 is a Wi-Fi network, then the network 120may include a Wi-Fi access point (AP) or Wi-Fi router 110 that managesthe network in infrastructure mode. Alternatively, the network 120 maybe using Wi-Fi Direct (Wi-Fi Direct is a standard of the Wi-FiAlliance), in which case the AP 110 may not be present. The IP network120 may also be connected to the internet 800 through a wide areanetwork connection 26. The source 104 may also have a remote device 114associated with it such as a remote control device connected via an IPor other communication link 116. In addition the source 104 or network120 may have additional optional devices 112 such as a NAS (NetworkAttached Storage) device that provides media.

IP networks can use several different types of messaging includingunicast, multicast and broadcast messaging. Messaging being the sendingof IP packets.

Unicast messaging is a type of Internet Protocol transmission in whichinformation is sent from only one sender to only one receiver. In otherwords, Unicast transmission is a one-to-one node transmission betweentwo nodes only. In unicasting each outgoing packet has a unicastdestination address, which means it is destined for a particulardestination that has that address. All other destinations that may hearthat packet ignore the packet, if the packet's destination address isnot the same as that destination's address. Broadcast is a type ofInternet Protocol transmission in which information is sent from justone computer, but is received by all the computers connected on thenetwork. This would mean that every time a computer or a node transmitsa ‘Broadcast’ packet, all the other computers can receive thatinformation packet. Multicast is a type of Internet Protocoltransmission or communication in which there may be more than one senderand the information sent is meant for a set of receivers that havejoined a multicast group, the set of receivers possibly being a subsetof all the receivers. In multicasting, each multicast packet isaddressed to a multicast address. This address is a group address. Anydestination can subscribe to the address and therefore can listen andreceive packets sent to the multicast address that it subscribed to. Thebenefit of multicasting is that a single multicast packet sent can bereceived by multiple destinations. This saves network traffic if thesame packet needs to be sent to multiple destinations. When the samedata needs to be sent to multiple IP destinations generally,Broadcasting or Multicasting, rather than Unicasting, provides the mostefficient use of the network.

In this description the terms Broadcast and Multicast may be used. Inboth Broadcasting and Multicasting, when messages are sent, they arereceived by multiple destinations. Therefore in the presentspecification, the terms Broadcast and Multicast may be usedinterchangeably to refer to one packet being received by multipledestinations. In some cases this description only says the media is sentor transmitted without specifying whether it is broadcast, multicast orunicast. In this case, it means any one of these methods may be used forsending or transmitting the media.

In this description, the terms Message and Packet are often used and maybe used interchangeably. A Packet is a data set to be sent or receivedon an Internet Protocol network. The Packet may or may not be the sameas an ‘Internet Protocol Packet’. A Message refers to the logicalinformation contained in such a packet. In this description, the termSegment may also be used to refer to a data set. A data set is a set ofbytes of data. Data may be any type of data, including media or controlor informational data. In this description the term data and packet mayalso be used interchangeable depending on context. Packet refers to adata set and data refers to data in general.

Many IP protocols are accessed from software programs via a Socketapplication programming interface. This Socket interface is defined aspart of the POSIX standard. POSIX is an acronym for “Portable OperatingSystem Interface”, which is a family of standards specified by the IEEEfor maintaining compatibility between operating systems.

Currently when the same media data needs to be sent to multiple networkdestinations, the general technique for doing so is to use datamulticasting to the multiple destinations that need to receive the data.

In such a system the media is multicast to all the destinations and itis up to each destination to attempt to render the media appropriately.If during rendering there is an error where a renderer does not receivenew media data or does not receive it correctly, the renderer may rendererroneous data and then attempt to recover and continue correct mediarendering from the point after the error when correct data is received.For example, during rendering of a H264 stream, if there is anincidental data drop out, the displayed image may pixilate briefly andthen recover.

In the applications envisioned here, there is a need to send media froma source to multiple media devices, such as TV and speakers in the samelistening and viewing space. Furthermore there is a need to send thismedia over a wireless network such as Wi-Fi.

For these applications, this means all of the media rendering devices,such as speakers, that are in the same listening or viewing zone, needto be precisely synchronized to each other, so the listener and/orviewer does not discern any unintended media experience.

Secondly, because the media is transported over wireless, there is avery high likely hood of a media error, where the media is not receivedat each destination reliably or uniformly. If using broadcast ormulticasts to send packets, the same broadcast or multi cast packet, maybe received at one destination but not received/heard by anotherdestination.

In order to synchronize the rendering of all media destinations, thisinvention uses a technique as described in U.S. patent application Ser.No. 11/627,957.

In this invention, in order to broadcast media over a Wi-Fi network, itis first necessary to recognize that broadcast or multicast media willnot be received at all destinations uniformly. Some destinations willreceive a multicast packet, while others will not.

IP networks were first designed to operate over wired networks. Bydesign, the packet communications on these networks were ‘best effort’.This means any packet transmitted on the network may not be received bythe intended destination. This is most often due to a collision, whereanother device starts to communicate at the same moment as the device ofinterest, thereby causing a collision. Another method of loss would bethe devices in the network path, such as routers, simply dropping thepacket, for example due to the lack of buffer space. Other reasons forloss could be that the wired line is simply noisy and the packettransmission got corrupted, though this is rare for the wired case vs.the wireless case.

In all these wired situations, it is generally the case, that if thetransmission, for example a multicast message, was received by onedevice on a ‘subnet’ or wire, all the other devices on the same ‘wire’or subnet also receive the transmission correctly. This is because inthe wired case, the noise or interference situation of a device on onepart of the wire is not so different from the noise situation at anotherpart of the wire. If the wired devices are connected via a switch ratherthan a hub, the same issues are true, the amount of noise orinterference is minimal.

In Wi-Fi the differences in receipt of Wi-Fi traffic at each Wi-Fidevice in a subnet is substantial. Therefore it is necessary to accountfor this in the applications described in this invention.

To account for the differences in Wi-Fi traffic receipt at each device,this invention proposes a mechanism, referred to as ‘Managed Receipt’where the media broadcaster manages the receipt of media packets at eachmedia receipt device. Note any reference to the word ‘broadcast’ refersto the English meaning of the word as well as the IP communicationmethods of both broadcasting and multicasting. Broadcasting may also beimplemented by unicasting IP packets to several destinations—which hasthe same effect as a broadcast, though it is not an efficient use of thenetwork, if the entire payload in each packet is the same.

Traditional Protocols

In traditional point to point transmission protocols such as TCP, theinternet standard Transmission Control Protocol as defined in InternetProtocol Request for Comment 793 (RFC 793), there is a need to haveguaranteed packet transmission from end to end. TCP does so, by storingthe packets to be sent in a buffer at the source, sending the packets tothe destination and then discarding the packets at the source only whenthe destination end acknowledges receipt of the packet. If thedestination does not acknowledge a packet, it is retransmitted.

FIG. 3 shows a typical TCP connection between a data source 104 and adata destination 106. The source 104 has an application 652 that usesTCP 658 for transporting media from the source 104 to the destination106. The destination 106 has an application 670 that receives data fromthe TCP 658 layer. The TCP 658 layer consists of a TCP source 656, a TCPover IP network link 660 and a TCP destination 662. The sourceapplication 652 sends data via the TCP layer 658 by using a socketinterface 654. The destination application 670 receives data via asocket interface 666.

TCP manages the flow of data by providing receipt information at the TCPdestination 662 to the TCP source 656 and the TCP source 656 using thisinformation and a flow control algorithm to control the flow of data.FIG. 4 shows a detailed view of the TCP source 656, which contains aretransmission queue 300 and a sliding window 308 that maintains currenttransmission status information.

FIG. 4 shows the sliding window 308 algorithm used by TCP to control theflow of data from the TCP source 656 to the TCP destination 662 (SeeFIG. 3). When a data set is sent from the source 656 (see FIG. 3) to thedestination 662 (see FIG. 3), the data is grouped into segments and eachsegment is marked with a sequence number that represents the byte numberof the first byte of the segment. During sending 654, an outgoingsegment 302 is sent 660 from the source 656 to the destination 662 andthen put 306 in a retransmission queue 300 at the TCP source 656. Asdata is sent from the TCP source 656 to the TCP destination 662 andreceived by the destination 662, the destination 662 sends acknowledgemessages back to the source 656 indicating the highest sequence numberthe destination 662 has received. Any segments in the retransmissionqueue 300 that have a sequence number that is equal to or less than thehighest sequence number the destination 662 has received is discarded314 by the source 656 from the retransmission queue 300. The slidingwindow 308 in the retransmission queue 300 therefore contains segments312-316 that have been sent to the destination, but have not beenacknowledged. When data is acknowledged by the destination, thedestination includes a sliding window 308 size 318 to use by the source.This sliding window 308 size 318 is used by the source 656 to limit thenumber of segments the source 656 can send that have not beenacknowledged.

For the earliest segment 312 that has not been acknowledged, the source656 keeps track of a timeout value. If this value is greater than theRetransmission Time Out (RTO), then the source 656 restarts transmittingsegments from this segment 312. The RTO value is computed based on acalculation using the round trip time. The RTO is dynamically computedand can change over time.

Some key points to note about the TCP protocol are that:

When TCP is sending data, it is sending to a single destination.

In TCP, any segment that is received over the network with a sequencenumber that is out of order and greater than the next sequence numberthe destination 662 is expecting is discarded by the destination 662.This discarding of data received over the network, just because it wassent out of order, is very wasteful of network resources.

In this invention, as described below, out of order packets are saved inthe receive queue for use later.

In TCP, after a segment times out, the TCP protocol restarts sendingfrom the beginning of the retransmit window. This means any segmentsthat have been received by the destination and have now been discardedare resent. This is again wasteful of network resources. In thisinvention retransmission of packets that have already been received isavoided.

In TCP the destination only indicates the highest consecutive sequencenumber it has received. In this invention both the highest consecutivesequence number and any missing packets are reported by the destination.

TCP philosophy is a slow start. This invention is tailored for a faststart and slows down fast when network conditions are bad.

TCP assumes that packet losses are occurring due to congestion andtherefore slows the throughput by reducing the amount of data in transit(reducing windows size 318) and waiting longer for acknowledges.However, in wireless networks, packet loss may occur unrelated tocongestion. So when TCP is used over wireless and it decreases the rate,due to packet losses rather than congestion, this rate decrease isinappropriate and is an undesired effect.

FIG. 5 shows a timeline of packet (data set/segment) transmission andreceipt at the source 400 and destination 402 using TCP. This shows aninitial set of packets 1 through 6 being sent from the source 400. Ofthis, an initial set of packets 404 marked 1, 2 and 3 are received atthe destination 402. The next set of packets 410, packets marked 4 and5, are not received at the destination 402 until time 416 and thereforethe acknowledge message for these packets 407 is not received by thesource 400 till time 418, which is well past the timeout period RTO 412.Therefore the source 400, when the timeout period RTO 412 expires, willretransmit packets 414 starting at the timed out packet marked 4 at time420. Depending on what the timeout period RTO 412 is set at and what thedelay in arrival of the initial set of packets marked 4 and 5 410, thesecond set of packets 414 may arrive before time 416 or after time 416.

A key point regarding this algorithm is that the decision to retransmitpackets is made by the source 400 based on a timeout period RTO 412 andthe lack of an acknowledge message from the destination. Therefore thesource 400 makes decisions based on limited information about receiptconditions at the destination 402. Secondly, because TCP assumes thoseacknowledge delays or acknowledge absence is largely due to congestionrather than interference, it errs on the side of a long RTO and smallerwindow size and increases and decreases these respectively when timeoutsoccur. When packets are merely lost due to interference, such as in thecase with Wi-Fi networks, rather than due to network congestion, thiscauses an inadvertent drop in transfer rate that is undesirable andunnecessary. Thirdly, because it restarts sending all packets in theretransmission queue, a significant amount of unnecessary repeat trafficcan occur, wasting network bandwidth.

During heavy packet transmission, the network stack and all devices inthe network path, such as routers and access points may have manypackets to transmit and receive from many sources. Since the order inwhich packets are processed at the IP layer does not have to be in thesame order as it is received for service, they may be handled in anyorder. Therefore as shown in FIG. 5, packets marked 4 and 5 sent in set410 may arrive much later than a retransmission of these packets sent inset 414. For example, packets 4 and 5 in set 410 may have been sent toan 802.11 access point from a source network adapter (uplink). Theaccess point may then process other packets from other network adaptersand may even lose RF connectivity with adapters. The access point maythen reconnect and receive a new set 414 of packets 4 and 5 from thesource network adapter and send them to the destination adapter(downlink), before finishing the downlink of packets 4 and 5 it receivedin set 410, and that was waiting in a queue for processing, by sendingthem to the destination network adapter.

FIG. 6A shows a plot of the number of packets received at various packetto packet receive delays measured over a period of time. The packet topacket receive delay is the time interval between the receipt of apacket and the receipt of the next packet. The Y axis 554 shows thenumber of packets received and the X axis 552 shows the delays. The plot550 shows that the packet to packet delays rise to a peak and then falluntil they reach a point 564 where the delay goes to infinity, i.e. thepacket is lost and will never arrive.

If a transmission algorithm waits for a timeout period T_(O) 560 forpackets to arrive before requesting retransmission of these packets andif T_(O) is set to be longer than the largest delay 564 seen, then thesystem will only retransmit packets that are truly lost. If the timeoutperiod is set to a shorter interval such as T_(a) 562, then on manyoccasions the system will retransmit packets that are still on its wayand may arrive later, after the timeout period T_(a) has expired.

FIG. 6B shows the effect of this in more detail. This is a plot 568 ofthe total number of packets to arrive under a delay of T, as a percentof the total number of packets sent over the measurement period. In thisplot, the Y axis 566 is the total packets received over the totalpackets sent as a percent and the X axis 567 is various possible timeoutperiods. If the timeout period is T_(O) 560 and the corresponding delayis 506, then the percent of packet retransmissions 508 will only be forlost packets. All other transmissions 504 will be for received packets.If the timeout period is set to T_(a) 562 a shorter interval, and thecorresponding delay is 559, then the percent of packet retransmissions558 will not only include lost packets 508, but also a significantamount of packets 561 that are already on the way, but are just slow.This means that as the timeout period is shortened, the amount of excessrepeats of packets already in transit increases. This retransmission ofpackets that have already been transmitted and will arrive in due courseis wasteful of network resources. However, a short timeout period willensure that packets that would otherwise be slow to arrive are receivedmore promptly, by requesting their retransmission. Therefore the choiceof the timeout period T is a tradeoff between requesting retransmissionof delayed packets to ensure prompt packet receipt and wasting networkresources in excessive retransmissions of the same packets.

When it comes to point to multi point transmission, current protocolssuch as RTP, the Real Time Transport Protocol, simply multicast allpackets to all the destinations. There is no mechanism to guaranteepacket transmission and receipt to all destinations. Each destinationgets what it gets. RTP therefore is effective in getting the same packetto multiple destinations, but is not helpful in ensuring all packets getto each destination.

Loss Anticipation and Preemption Algorithm

In many applications, as in FIG. 1, there is a need to get media fromthe media source 104 to the media destination 106 as promptly aspossible, i.e. with minimum delay or latency. In addition, there is aneed to get all the data to the destination with no data losses. Thesetwo requirements are conflicting requirements. If the underlyingtransport can lose packets, then in order to recover from data loses, itis necessary to retransmit these packets. Doing retransmission howevertakes time and so increases overall latency. Note it is possible to doforward error correction in some cases, but when whole packets can belost, it is more effective to retransmit the lost packet.

For example, the destination 106 may be rendering media sent from themedia source 104 at a certain time offset from the time it was sent fromthe media source 104. If the media takes longer than this time offset toget to the destination 106, then the destination 106 will run out ofmedia data to render and will underflow. Similarly, if the data sent tothe destination 106 get lost along the way, then the destination willhave no media to render, when it comes time to render the media that waslost and rendering will be erroneous. Therefore it is necessary toensure that all media is received at the destination and if anyretransmission needs to take place that it takes place and lost data isreceived before the destination rendering underflows.

The time lost in retransmission includes the time it takes for a packetto be identified as lost at the destination, a loss notification beingsent to the source, the source resending the packet and theretransmitted packet being received by the destination. This total timeadds to the total system ‘latency’. If it takes 100 mSecs for thisprocess to occur, data receipt at the destination will be delayed by 100msecs.

In order to keep this latency low, it is necessary to keep these delaysas low as possible. In general the actual physical transmission timesare low, in that they are not much more than the packet transmissiontimes, measured in 1-2 mSecs.

A major delay in the retransmission process is how long it takes todetect that a packet is lost.

This invention describes a process/algorithm known as a LossAnticipation and Preemption algorithm (LAP) for keeping this lossdetection delay low. This LAP algorithm is implemented in a transportprotocol named FCP (FireCast Control Protocol).

This LAP algorithm is designed to make early pre emptive decisions aboutpotential packet loss and have these packets retransmitted. This, asmentioned above, is a tradeoff of using additional network resources toensure the more timely availability of packet data.

In addition, unlike in TCP where the TCP source makes the decision toretransmit data, in LAP it is the destination that makes the decision torequest a retransmission of data. The destination has more accurate andup to date information regarding the status of data received and istherefore in a better position to make timely requests forretransmission.

FIG. 7 shows what can happen in a typical Wi-Fi network. In general aWi-Fi network will transmit the packets from Source 104 to destination106 in the order that they were given to the source for transmission.However, in transmission, these packets may be lost, the order of packettransmissions mixed up and each packet may be delayed by an unknownamount of time. The occurrence of these is determined not only beRF/Wi-Fi factors as described above, but also due to implementationissues, such as buffer sizes in the access point and network trafficcongestion at that particular moment. In FIG. 7 a set of packets 1through 6 are sent from the source 104. Of these, packets 1 through 3are received 420 after a transit time at the destination 106. Packets 4through 5 are delayed and arrive 422 at the destination 106 with asignificant delay. Packet 6 however is not delayed and arrives 424 atthe destination after a transit time.

In a traditional TCP type sliding window algorithm, the source 104 wouldwait a timeout period TT 430 for an acknowledge for the packets it sentand then would retransmit packets 4 through 7

However, if the destination 106 were to detect on receipt of packet 6that packets 4 and 5 were missing and if it were to use this informationand a short timeout period to request retransmission of these packetsfrom the source 104, it is possible to get these packets to thedestination 106 before the delayed packets 422 finally arrive. This iswhat the LAP algorithm in this invention does. In this invention, ametric is computed for the chance of a packet that has not been receivedas being lost. This metric is based on multiple factors; the order ofreceipt; delay in receipt and need for data expressed by an applicationusing the media data. This metric is then used to request 426retransmission of packets, as shown in FIG. 7, and receive these packets428 before the delayed packets 422 arrive.

FIG. 8 shows the overall architecture used. The system consists of amedia data source 104 and a media destination 106. The media source hasa media application 624 that sends media to a media destinationapplication 620 that is in the media destination 106. In thisapplication, as mentioned above, it is desirable to get the media fromthe source application 624 to the destination application 620 both withas low latency as possible and with no data loss.

As in the TCP case, the media application provides the media to betransported to a transport layer 604, named the FCP (FireCast Protocol)layer, via a transport layer interface 606. The FCP transport protocolconsists of a FCP source 622 and a FCP destination 618 and an IP networklink connecting them. This network link sends data packets 610 to theFCP destination 618 and status messages back 612 from the destination618 to the FCP source 622. The status messages 612 include informationalmessages and request to retransmit packets. The destination application620 receives the media data via an application interface 616 to the FCPtransport layer 604.

Unlike in the TCP case, in FCP the destination application 620 that isthe consumer of the data provided by the FCP transport, in addition toreceiving data through the application interface 616 also indicates tothe FCP transport layer its current level of urgency or need for data tothe Transport Layer 604.

In addition, unlike in TCP, the FCP destination 618 is not a passiveprovider of information to the source. In FCP, the FCP destination 618makes decisions regarding the data packets it has received and makesdata packet retransmission requests 612 to the FCP source 622. The FCPsource 622 services these requests. Data packet timeout decisions aremade at the destination 618.

FIG. 9 shows this system in more detail. The destination application 620is the consumer of media data packets 616 coming from the media source624 (in FIG. 8) via the transport layer 604. The media rendering mediadestination application 620 has a buffer 635 to store the media data ithas received via the transport interface 616 and has a media renderingcomponent 630 that takes media data from the buffer and renders it 632.

The destination applications 620 using FCP also provides informationindicating its current Need level 642 to the FCP transport interface614. This Need level indicates the destination/media data consumerapplications need for data to the transport level 604 and is computed ina variety of ways. In this embodiment as the amount of media dataavailable in the buffer 635 gets low, the Need for media data gets high.This Need value is computed by logic 634 that is monitoring the level636 of valid media data in the buffer 635.

The FCP destination 618 includes a received packet collection (RPC) 320and LAP algorithm process 636 to provide receipt status and packetretransmission requests 612 to the FCP source 622 (see FIG. 8). The LAPalgorithm 636 periodically scans 644 the RPC 320 together with the needlevel 614 to make the retransmission requests 612.

Each packet that is transmitted across the network is given a uniqueidentifier by the source 622 (see FIG. 8), a PID (Packet Identifier).Packets are therefore identified by their PID and may be referred to asa PID in the description below. Since the destination 618 may receivethese packets out of order, it has a reordering mechanism built into it.In this mechanism, the packets 648 are put INTO the received packetcollection (RPC) 320 on receipt, in any order, as they are received. Thepackets are taken OUT 638 of the RPC 320, based on PID order. I.e.packets are removed in incrementing PID order. If the next PID is notpresent in the RPC 320, the destination's 618 packet removal processhalts, waiting for this PID.

The goal of the LAP algorithm is to ensure that the next PID required isalready always present in the RPC 320 at the destination, when thedestination application 620 wants to use/consume it by moving it to itsbuffer 635.

Since the packets in the RPC are received in any order, the contents ofthe RPC 320, see FIG. 10, can be viewed as a linear list of Packets with‘gaps’ 322, 323, 324 in the list for packets that have not yet beenreceived. These packets 322, 323, 324 that have not been received yetare identified by their PIDs, and are referred to as missing PIDs 322,323, 324.

FIG. 10 shows details of the FCP destination 618 shown in FIG. 8 andFIG. 9. Some of the references in this description refer back to FIG. 8and FIG. 9. In FIG. 10 the latest packet received 610 from the FCPsource 622 (see FIG. 8) is 316 with PID 27. The oldest and next packetthat the destination application (see FIG. 8) 620 may consume via link638 is 309 PID 04. It shows PID 16, 322, PID 20 323 and PID 21 324 asmissing. Associated with the RPC 320 is a media In-Transit window 310that identifies the part of the RPC 320 that has missing PIDs in it. TheIn-Transit windows starts 311 from the oldest missing PID location 322,PID 16 in this diagram, and ends 302 at the most recent PID, 316, PID27. When the oldest missing PID 322, PID 16 is received, the start 311of the In-Transit window will move to PID 20, 323 in this diagram, ifPID 18 is received before PID 20. The width of the In-Transit window310, which starts at the oldest missing PID 322 and ends at the mostrecent PID 316, is Twd 325. The amount of PIDs available Tavail 327 inthe RPC 320 includes the oldest packet received 309 to the packet 307just before the oldest missing packet 322. These PIDs are available tobe used by the consumer application 620.

The LAP algorithm periodically scans the RPC 320 and identifies PIDS inthese ‘gaps’ as missing PIDs 322, 323, 324. A missing PID is a packetthat has not been received, but has another packet received after itwith a higher PID, than the missing PID.

For each such missing PID, the LAP also computes a metric based on howlate the missing PID is with respect to the previous packet received.

Based on this metric, the LAP then sends a list of missing packet PIDsto the FCP source 622 (see FIG. 8). The net effect of this is that thedestination 618 asks the FCP source 622 (see FIG. 8) to retransmitpackets that are likely lost in the system, before the destination 106(see FIG. 8) actually needs the packets.

The cost of this method is that the receiver may identify packets asmissing that are actually still in transit and may be subsequentlyreceived. This amounts to some degree of duplicate transmissions. TheLAP adjusts its missing PID identification scans and late packet metricto meet a balance between duplicate transmissions and minimal delayswaiting for missing packets to be retransmitted.

In this embodiment the missing PID identification scans are performedperiodically at a 20 msec rate. A longer period may be used to reduceprocessor load for applications that do not require short latency. Forvery short latencies this scan period can be set as low as 5 msec tocheck for missing packets very frequently.

In some cases the missing packet is the last packet in a sequence andhas no following packet. In these cases, the LAP will trip a basic delaytimeout and identify the packet as a late last packet based on this.

FIG. 11 shows the LAP algorithm used by the destination to requestpacket retransmissions in detail. The LAP process starts 900 by 902setting the working pointer to the packet at the start 311 (see FIG. 10for some of the following references) of the In-Transit Window 310 andthen checks this packet position 906. If in checking for being the lastpacket 908 this is not the end of the In-Transit Window 302 and inchecking for a missing packet in step 910 a packet is not present atthis position 910, this is a missing packet and it moves to calculate aD_(m) value in step 914. The D_(m) value is calculated using the delaysince the previous most recent packet that was present and a currentNeed level 904 provided by the destination application 620 (see FIG. 8for some of the following references) consuming these media datapackets. See calculations below for computing D_(m) and the followingvalues. With D_(m) a missing packet metric M_(m) is computed 916 andthis metric is compared to a threshold 922. If above the threshold 922,a retransmit priority R_(m) is calculated 926 and this missing PID isadded 928 to a list of Late/Missing PIDs that will be requested forretransmission by the destination 618. If M_(m) in step 922 is below thethreshold, the working pointer is moved to the next packet 930 and thispacket position is checked 906. If in step 910 the packet is present, noaction is taken and the working pointer is moved to the next packet 930and this packet position is checked 906.

If in checking if this is a last packet 908 the working packet positionis at the end of the In-transit Window 302, then this is the last packetposition 908. A D_(l) value is calculated 912 for this last packet usingthe delay since the previous most recent packet that was received and acurrent Need level 904 provided by the destination application 620consuming these media data packets. See calculations below for computingD_(l). Then a last packet late metric M_(l) is computed 918 and thismetric is compared to a threshold 920. If above the threshold 920, aretransmit priority R_(l)is calculated 924 and this late PID is added934 to a list of Late/Missing PIDs that will be requested forretransmission by the destination 618. In either case, since the workingpointer is at the end of In-Transit window the LAP process ends 932.

Below are the calculations performed in the LAP processes describedabove.

DEFINITIONS:

-   T_(lp): Time previous packet was received-   T_(c): Current time-   d_(m): Delay of Missing packet=T_(c)−T_(lp)-   D_(thm): Delay Threshold of Missing Packet-   D_(minm): Delay Minimum for Missing Packet-   D_(m): Delay as a percent of thresh hold-   M_(m): Metric for Missing Packet as a percent-   N: Need as a percent-   K_(m): 1/16-   d_(l): Delay of Last packet=T_(c)−T_(lp)-   K_(l): 1/16-   D_(thl): Delay Threshold of Last Packet-   D_(minl): Delay Minimum for Last Packet-   D_(l): Delay as a percent of thresh hold-   M_(l): Metric for Last packet as a percent-   MIN(A, B): returns the minimum of A or B-   R_(m): Missing Packet retransmit priority-   R_(l): Last Packet retransmit priority-   D_(x): D_(m) or D_(l)-   M_(x): M_(m) or M_(l)-   R_(x): R_(m) or R_(l)

Missing Packet Metric Calculations

D _(m)=(d _(m) −D _(minm))/D _(thm)

M _(m) =F _(n)(N)*F _(m)(D _(m))*K _(m)

F _(n)(N)=2̂(N/8)

F _(m)(D _(m))=2̂(D _(m)/8)

M _(m) =MIN((( 1/16)*2̂((N/8)+(D _(m)/8))), 100)

R _(m) =N

Last Packet Metric Calculations

D _(l)=(d _(l) −D _(minl))/D _(thm)

M _(l) =F _(n)(N)*F _(l)(D _(l))*K _(l)

F _(l)(D _(l))=2̂(D _(l)/8)

M _(l)=MIN((( 1/16)*2̂((N/8)+(D _(l)/8))), 100)

R _(l) =N

FIG. 12 shows these calculations computing M_(x)(M_(m) or M_(l)) forvarious Needs, N, vs various Delays D_(x)(D_(m) or D_(l)) for thresholdsD_(thm) or D_(thl) set to 100 and K_(x)= 1/16. FIG. 13 shows a plot ofthese numbers. As can be seen as the Delay gets closer to 100% or theNeed gets closer to 100%, the Metric reaches 100%. When the FCPdestination 618 send a list of Late/Missing PIDs to the FCP source 622,it includes for each Late/Missing PID the metric M_(m) or M_(l) and theretransmission priority R_(m) or R_(l) for that packet. The source canuse this information to adjust how it processes and sends packets toeach destination.

The present invention has been described in particular detail withrespect to several possible embodiments. Those of skill in the art willappreciate that the invention may be practiced in other embodiments.First, the particular naming of the components, capitalization of terms,the attributes, data structures, or any other programming or structuralaspect is not mandatory or significant, and the mechanisms thatimplement the invention or its features may have different names,formats, or protocols. Further, the system may be implemented via acombination of hardware and software, as described, or entirely inhardware elements. Also, the particular division of functionalitybetween the various system components described herein is merelyexemplary, and not mandatory; functions performed by a single systemcomponent may instead be performed by multiple components, and functionsperformed by multiple components may instead be performed by a singlecomponent.

Some portions of above description present the features of the presentinvention in terms of methods and symbolic representations of operationson information. These descriptions and representations are the meansused by those skilled in the data processing arts to most effectivelyconvey the substance of their work to others skilled in the art. Theseoperations, while described functionally or logically, are understood tobe implemented by computer programs. Furthermore, it has also provenconvenient at times, to refer to these arrangements of operations asmodules or by functional names, without loss of generality.

Unless specifically stated otherwise as apparent from the abovediscussion, it is appreciated that throughout the description,discussions utilizing terms such as “determining” or the like, refer tothe action and processes of a computer system, or similar electroniccomputing device, that manipulates and transforms data represented asphysical (electronic) quantities within the computer system memories orregisters or other such information storage, transmission or displaydevices.

Certain aspects of the present invention include process steps andinstructions described herein in the form of a method. It should benoted that the process steps and instructions of the present inventioncould be embodied in software, firmware or hardware, and when embodiedin software, could be downloaded to reside on and be operated fromdifferent platforms used by real time network operating systems.

The present invention also relates to an apparatus for performing theoperations herein. This apparatus may be specially constructed for therequired purposes, or it may comprise a general-purpose computerselectively activated or reconfigured by a computer program stored on acomputer readable medium that can be accessed by the computer. Such acomputer program may be stored in a tangible computer readable storagemedium, such as, but is not limited to, any type of disk includingfloppy disks, optical disks, CD-ROMs, magnetic-optical disks, read-onlymemories (ROMs), random access memories (RAMs), EPROMs, EEPROMs,magnetic or optical cards, application specific integrated circuits(ASICs), or any type of media suitable for storing electronicinstructions, and each coupled to a computer system bus. Furthermore,the computers referred to in the specification may include a singleprocessor or may be architectures employing multiple processor designsfor increased computing capability.

The methods and operations presented herein are not inherently relatedto any particular computer or other apparatus. Various general-purposesystems may also be used with programs in accordance with the teachingsherein, or it may prove convenient to construct more specializedapparatus to perform the required method steps. The required structurefor a variety of these systems will be apparent to those of skill in theart, along with equivalent variations. In addition, the presentinvention is not described with reference to any particular programminglanguage. It is appreciated that a variety of programming languages maybe used to implement the teachings of the present invention as describedherein.

The present invention is well suited to a wide variety of computernetwork systems over numerous topologies. Within this field, theconfiguration and management of large networks comprise storage devicesand computers that are communicatively coupled to dissimilar computersand storage devices over a network, such as the Internet, publicnetworks, private networks, or other networks enabling communicationbetween computing systems.

The applications this invention are directed at that may be describedabove and any objects of this invention that are described above do notfully describe all the applications and objects of this invention andthese descriptions are not intended to be limiting in anyway or manner

Finally, it should be noted that the language used in the specificationhas been principally selected for readability and instructionalpurposes, and may not have been selected to delineate or circumscribethe inventive subject matter. Accordingly, the disclosure of the presentinvention is intended to be illustrative, but not limiting, of the scopeof the invention, which is set forth in the following claims.

1. A system for low latency transport of media data over an IP networkfrom a media source to media destination devices, comprising: Aninternet protocol communication network; a media source devicecomprising a media transport source component and having a means forcommunication with said communication network; a media destinationdevice comprising a media transport destination component and a mediaconsuming component and having a means for communication with saidcommunication network; and the said media transport source componentsending media data in packets to the said media transport destinationcomponent using the said communication network; the said media transportdestination component receiving the said packets and placing them in acollection and using the said packets in the said collection to providemedia data to the said media consuming component; the said mediatransport destination component detecting missing packets in the saidcollection and computing a time delay for each said missing packet andusing this time delay to make a decision whether to ask the said mediatransport source component to retransmit the said media packet and thesaid media transport destination sending requests for the retransmissionof missing packets to the said media transport source.
 2. The saidsystem of claim 1 where the said media transport source componentidentifies each said packet sent with a sequentially incrementing packetidentifier and The said media transport destination component detectingmissing packets done by noting the packet with the highest said packetidentifier that has been received and detecting any packet that has notbeen received with a packet identifier that is lower than this saidhighest packet identifier as a missing packet.
 3. The system of claim 1wherein the said decision consists of computing a retransmission metricfor the said missing packet and comparing this retransmission metric toa threshold and deciding to ask for the said packet to be retransmittedif this retransmission metric is above the said threshold.
 4. The systemof claim 1 wherein the time delay for the said missing packet iscomputed by subtracting from the current time, the time that the latestpacket, that has been received, with a packet identifier that is lowerthan the said missing packets packet identifier.
 5. The system of claim3 wherein the said media consuming application provides a media dataneed value to the said media transport destination component and Thesaid media transport destination component using this said need value incomputing the said retransmission metric for each said missing mediapacket.